Information technology (IT) and Communication Networks (Networks) provide end users with access to communications and diverse media. Typical Networks include, for example: PSTN (Public Switched Telephone Network), IP-based (Internet Protocol, or Internet), ISDN (Integrated Services Digital Network), and PLMN (Public Land Mobile Network). Networks provide end users with diverse applications such as: wired and wireless voice, multimedia access/transmission, Internet access, Voice over IP (Internet Protocol), etc. Access to and communication over Networks is accomplished via hardware (e.g, switches, routers, servers, gateways, gateway controllers, etc.). and communications protocols (e.g., IP (Internet Protocol), RLP (Radio Link Protocol), ATM (Asynchronous Transfer Mode), GSM (Global System for Mobile Communication), CDMA (Code Division Multiple Access), TDMA (Time Division Multiple Access), RTP ( ), SS7 (Signaling System 7), SIP (Session Initiation Protocol), MGCP (Media Gateway Control Protocol), etc.).
A Network link may generally be set up by using access technology, such as PRA (Primary Rate Access) signaling or ISUP (ISDN user part). Functionality to set up links between, for example, a Media Gateway (MG) and the Internet, normally reside on a separate hardware component such as an Access Server (AS). Once the AS gets signaled by the Mobile Switching Center (MSC), an incoming communication requesting access is directed from the MSC towards the Media Gateway Controller (MGC). Once the MGC gets signaled via a communication requesting Internet access, the MGC seizes AS functionality.
Internet applications such as IA (Internet Access), VoIP (Voice over IP) and Multimedia generally require certain functionality such as contained within a MGC to control the media stream in the MG as well as other Internet specific functionality such as AAA (Authentication, Authorization, Accounting).
Networks access and operational inefficiencies generally occur because of bandwidth limitations or hardware clashes. For example, a traditional telephone call is established over a dedicated network circuit (e.g., Public Service Telephone Network, or “PSTN”). Regardless of the amount of voice data (e.g., analog voice signals) to be transferred across the circuit connection (e.g., via the PSTN), the entire circuit must generally be dedicated for the duration of the call; thus resulting in a potentially low overall usage of switching resources. An alternative to circuit switched (voice) calls is packet switched calls (e.g., voice over IP). In a packet switched call, the voice data is encoded into packets which are then directed across a network and reassembled into an approximation of the original data at the receiving end. During a silent period of a call, no data generally needs to be packetized and transmitted across the network; thus saving network resources and allowing for higher call density (e.g., more calls can be routed on the same amount of network resources).
In order for a call to be sent over a packet-switched network, the voice data (e.g., analog voice signals) must be sampled and converted into a digital format (e.g., packetized voice). The sampling process introduces a trade-off between sound quality and bandwidth utilization. A voice stream can generally be sampled very frequently to produce very high sound quality on the receiving end, but at the cost of using a large amount of bandwidth to transport generated packets. Alternatively, voice can be sampled infrequently to produce a smaller amount of packet data but at the cost of low sound quality at the receiving end.
As part of setting up a call over a packet network, a codec (coding/decoding algorithm) must generally be specified to control what rate of sampling is used to encode the call. Depending on the codec algorithm specified, various levels of voice quality/bandwidth utilization can be generally achieved.
There is apparently a lack of available methods and systems for a caller making a call on a voice over IP network to specify, during a call, a level of voice quality or Quality of Service (QoS) that is desired. While the network operator can generally specify at initial call setup or even in mid-call what quality level will be used, the caller apparently cannot. Such a lack of user control over QoS precludes the network operator from offering low Quality of Service (QoS) at a low cost while still allowing the caller to upgrade to a higher QoS at a higher cost as needed during, or for the remaining duration of, a voice communication.